1. Field of the Invention
The present invention relates to an audio signal encoding method, an audio signal decoding method, an encoding device, a decoding device, an audio signal processing system, an audio signal encoding program, and an audio signal decoding program.
2. Description of the Related Art
A coding technique for compressing speech/music signals (audio signals) at low bit rates is important to reduce the costs incurred in communications, broadcasting, and storing of speech and music signals. In order to efficiently encode both speech signals and music signals, a hybrid-type coding scheme is effective in which a coding scheme suitable for speech signals and a coding scheme suitable for music signals are selectively utilized. The hybrid-type coding scheme performs coding efficiently by switching coding schemes in the process of coding an audio sequence, even when the characteristics of input signals vary temporally.
The hybrid-type coding scheme typically includes, as a component, the CELP coding scheme (CELP: Code Excited Linear Prediction Coding) suitable for coding speech signals. Generally, in order to encode a residual signal obtained through application of a linear predictive inverse filter to an input signal, an encoder exercising the CELP scheme holds therein information about past residual signals in an adaptive codebook. Since the adaptive codebook is used for coding, a high coding efficiency is achieved.
A technique for coding speech signals and music signals is described, for example, in Patent Literature 1. In Patent Literature 1, a coding algorithm for coding both speech signals and music signals, etc. is described. The technique described in Patent Literature 1 utilizes a Linear Predictive (LP) synthesis filter functioning commonly to encode speech signals and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator according to whether a speech signal or music signal is coded, respectively. For coding speech signals, the conventional CELP technique is used, and for coding music signals, a novel asymmetrical overlap-add transform technique is applied. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted on a signal in overlap-add operation regions.
When switching takes place from a coding scheme other than the CELP coding scheme to a coding scheme exercising the CELP scheme in the process of coding an audio sequence, information on a residual signal corresponding to the speech coming before the switching is not held in an adaptive codebook in the encoder. Therefore, the coding efficiency degrades when coding a frame coming immediately after the switching of the coding scheme, resulting in a problem of degradation in the reproduced speech quality. Conventional art is known such as Adaptive MultiRate Wideband plus (AMR-WB+, Non Patent Literature 1), which is a speech coding scheme standardized by the 3rd Generation Partnership Project (3GPP), in which the internal state of an encoder exercising the CELP scheme is initialized, using an encoded result obtained under a coding scheme other than the CELP scheme. The AMR-WB+ encoder obtains a residual signal through the linear predictive inverse filtering on an input signal and thereafter encodes the residual signal selectively using two coding schemes, i.e., the CELP scheme and the Transform Coded Excitation (TCX) scheme. When switching from the TCX scheme to the CELP scheme, the AMR-WB+ encoder updates the adaptive codebook in the CELP scheme, using an excitation signal in the TCX scheme.